Making peerconnection_unittest use the virtual socket server it creates.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1010 matching lines @@
   // removed in the received SDP.
   bool remove_cvo_ = false;
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
   std::unique_ptr<MockDataChannelObserver> data_observer_;
 };
 
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
-      : network_thread_(rtc::Thread::CreateWithSocketServer()),
-        worker_thread_(rtc::Thread::Create()),
-        pss_(new rtc::PhysicalSocketServer),
+      : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
-        ss_scope_(ss_.get()) {
+        network_thread_(new rtc::Thread(ss_.get())),
+        worker_thread_(rtc::Thread::Create()) {
     RTC_CHECK(network_thread_->Start());
     RTC_CHECK(worker_thread_->Start());
   }
 
   bool SessionActive() {
     return initiating_client_->SessionActive() &&
            receiving_client_->SessionActive();
   }
 
   // Return true if the number of frames provided have been received
@@ skipping 260 matching lines @@
   // Set the |receiving_client_| to the |client| passed in and return the
   // original |receiving_client_|.
   PeerConnectionTestClient* set_receiving_client(
       PeerConnectionTestClient* client) {
     PeerConnectionTestClient* old = receiving_client_.release();
     receiving_client_.reset(client);
     return old;
   }
 
  private:
-  // |worker_thread_| is used by both |initiating_client_| and
-  // |receiving_client_|. Must be destroyed last.
+  // |ss_| is used by |network_thread_| so it must be destroyed later.
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
+  // |network_thread_| and |worker_thread_| are used by both
+  // |initiating_client_| and |receiving_client_| so they must be destroyed
+  // later.
   std::unique_ptr<rtc::Thread> network_thread_;
   std::unique_ptr<rtc::Thread> worker_thread_;
-  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
-  std::unique_ptr<rtc::VirtualSocketServer> ss_;
-  rtc::SocketServerScope ss_scope_;
   std::unique_ptr<PeerConnectionTestClient> initiating_client_;
   std::unique_ptr<PeerConnectionTestClient> receiving_client_;
   bool prefer_constraint_apis_ = true;
 };
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 #if !defined(THREAD_SANITIZER)
 
 // This test sets up a Jsep call between two parties and test Dtmf.
@@ skipping 878 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace